It is often necessary for telecommunications network carriers to guarantee (or at least to be able to measure) a Quality-of-Service (QoS) level to (or for) its customers. One important aspect of a QoS measure is the total transmission delay—i.e., the delay from the speaker's mouth to the listener's ear, or equivalently, from the speaker's microphone to the listener's loudspeaker—across the network. Identifying problems with excessive transmission delay becomes particularly important when the network includes a packet-based IP (Internet Protocol) network—that is, where the telecommunications system comprises a Voice-over-IP (VoIP) design, where delays can be highly variable and traffic dependent.
Specifically, reliable estimates of total transmission (i.e., end-to-end) delays are often needed by telecommunications service providers for a number of reasons, such as, for example, (1) to perform a general assessment of network health for long-term provisioning and management, (2) to perform active call monitoring to ensure proper network operation, and possibly (3) to guarantee any QoS obligations made to end users, including that of a single “toll quality” category (which is typically made to all users). In addition, delay measurements are often needed by telecommunications equipment providers (4) to guarantee that contractual obligations are being met with respect to network performance using either existing standards such as the “e-model” (ITU-T/G.107) or other such similar devices, or (5) to determine delay budgets either dynamically during use or at design time. (ITU-T/G.107, which is also known as the “e-model,” is a well known standard promulgated by the International Telecommunications Union standards body and is fully familiar to those of ordinary skill in the art.) Additionally, (6) software products can be designed to allow dynamic adjustment of QoS parameters by assessing delay at the end-point. For example, when a PC-based (Personal Computer based) telephony application can assess delay on a per-call basis, it can then trade off delay requirements for bandwidth by adjusting the packetization rate on its transmitting channel and/or trade off packet loss for delay on its receiving channel.
Currently, there are three general methods for assessing the transmission delay which have typically been employed:
1) In a VoIP network design, IP header information may be used to calculate delay on the IP portion of the network. However, such techniques cannot assess the majority of various delay components that make up the total end-to-end delay in a complete system and as such, are not suitable for use in a QoS scheme. (This is true even for IP-terminal to IP-terminal networks.) Note that in many cases, the IP network contributes less than 10% of the overall delay, including the critical jitter buffer delay necessary on all VoIP calls.
2) A test signal or “probe” may be sent across the network. However, the use of this approach adds traffic to the network, and, moreover, it cannot measure the delays on actual customer connections. In addition, both ends of the network are often not under common service provider control—that is, one would need to control the terminals to get an accurate picture of the delay introduced by these devices, and it is typically not possible for a given service provider to introduce such a probe at an arbitrary terminal. Even if it were, probe-based measurements are insufficient because call delay cannot be determined for any arbitrary channel during any arbitrary time, and cannot account for variation in delay from terminal to terminal (which may in many cases account for the majority of total delay).
3) Recommendation ITU-T/P.561 suggests that “double-talk”—the situation in which both parties in a conversation talk simultaneously—can be used as an indicator of the existence of unacceptable delay. (ITU-T/P.561 is a recommendation promulgated by the International Telecommunications Union standards body and is fully familiar to those of ordinary skill in the art.) This results from the recognition that when excessive transmission delays are present, people naturally tend to talk over one another. Unfortunately, this approach provides merely a “litmus test” or true/false test for whether the transmission delay exceeds some threshold of acceptability (i.e., that which results in double-talk), and does not provide an indicator of the amount of delay. In addition, most echo cancellers, provided in many telecommunications network environments, interfere with (i.e., prevent the occurrence of) double-talk by switching to a half-duplex transmission mode when double-talk is detected.
Therefore, it would be highly advantageous if the total transmission delay across a telecommunications network used for speech transmission could be estimated without the limitations or disadvantages of the prior art techniques.